Troubleshooting Audio Quality
Problems in the Field
APP
NOTE
Issue 1.0
February 2006
Abstract
This document provides troubleshooting procedures for
resolving audio quality issues in the field.
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Table of Contents
1.
Introduction
2.
System Status and Environment
3.
Problem Definition
4.
Data Gathering
5.
Troubleshooting
6.
Audio Quality Anomalies
7.
Other Information
8.
References
APPENDIX A: Audio Processing Components & Audio
Quality Terminology
APPENDIX B: Basic IP Telephony Solution Diagram
3
4
5
5
6
7
11
12
13
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1. INTRODUCTION
Audio quality issues experienced by customers have
many and varied causes. The converged voice and data
solutions being deployed today require additional
consideration when troubleshooting these issues as
compared to the legacy voice solutions of the past.
Some problems may be caused by equipment configuration
or administration, others by network or environmental
conditions, some by user misuse, and some are simply a
customer expectation of the solution’s operation as
opposed to the solution’s inherent design intent.
This document provides a guide to troubleshooting a
customer’s audio quality issue(s) and determining what
component(s) in the call topology is causing the
problem. This is accomplished by first analyzing the
customer system and environment, and then defining the
customer’s audio quality issue using an industry
accepted definition of the anomaly and the typical
causes of that anomaly to thereby identify the area of
the customer’s solution (or network) in which to
concentrate the troubleshooting effort.
Appendix A provides a glossary of audio processing
components and audio quality terminology. Appendix B
provides a sample basic IP Telephony solution diagram
as a reference.
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2. SYSTEM STATUS AND ENVIRONMENT
As part of the normal Services troubleshooting
routine, it is necessary that system components are up
to date with the correct software/firmware releases,
hardware versions, etc. Once the system components are
established as having the latest updates, it is highly
recommended that Best Practices documentation be
consulted1. Areas not meeting the initiatives of Best
Practices documentation should be corrected and the
audio problem retested before proceeding in this
document.
Environmental conditions may cause or contribute to
audio quality problems. Local noise sources such as
computer fans, PC speakers, noise from air
conditioning (HVAC), ambient speech or noisy office
environments (as in call centers); can have an affect
on audio quality being perceived by the user when
using handset, headset or speakerphone.
For optimum speakerphone performance, make sure the
phone is sufficiently clear of any shelves or clutter
near the phone that could provide an echo path during
speakerphone use. Note that environments with hard and
reflective surfaces (such as metal walls and glass
windows) can promote reverberation when speakerphones
are used and result in echo complaints (from the farend).
1
Best Practices documents are intended for Avaya Services
Installation teams and Business Partners, and can be found on
the Avaya Enterprise Portal Support page via the Anatomy of a
Successful Cut link:
https://enterpriseportal.avaya.com/ptlWeb/appmanager/ptl/internal/?_nfpb=true&_pageLabel=support.
The link provides documents such as the Echo Prevention
Checklist which can be obtained by clicking on Checklists/FAQs
and then CSD – CM.
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3. PROBLEM DEFINITION
Assuming the system components are up to date (latest
version HW, SW, FW) and the solution design meets
Avaya’s IP Telephony deployment requirements2, the
audio quality complaint from the customer should be
translated into an industry accepted definition of
audio or voice quality related terminology as
appropriate. Appendix A provides a glossary of these
terms.
4. DATA GATHERING
Beyond the standard set of component model and release
information, gather additional meaningful information
such as call type (station to station or PSTN trunk
call), call traversal and solution topology (within
port network, across port network, LAN, WAN, etc.).
Information gathering on whether the user experiences
the anomaly on handset, headset, or speakerphone; or
whether any operation is being performed (transfer,
conference, etc.) when the anomaly occurs can help in
determining the cause of the anomaly.
2
Avaya solution deployment guidelines and network requirements are contained in document ID
555-245-600 or can be accessed at the following URL:
http://support.avaya.com/elmodocs2/comm_mgr/r3/pdfs/245600_3_4_1.pdf
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5. TROUBLESHOOTING
With the information gathered (as described in section
4), it is important that you outline the exact call
flow or the end-to-end route the voice signal takes
from the service provider network (if applicable)
through the system and across the customer network to
the end network device. Appendix B provides a network
diagram with measurement points identified (A – D) for
troubleshooting an incoming PSTN call to an enterprise
user. The following paragraphs are meant to assist in
determining the call flow.
If the audio quality problem involving Legacy
Telephones is intermittent, the support engineer will
be required to be available onsite or remotely at the
time the anomaly occurs (this may require a pager or
other notification process be established between
customer and support engineer so that the trouble can
be addressed in real time).
If the audio quality problem involving IP Telephones
is intermittent, a network analyzer (such as Agilent)
should be set up for 24/7 operation such that the call
information is captured (and buffered) at the time the
anomaly occurs. The key to ensuring the anomaly is
captured is by stopping the trace within a short time
after the problem occurs. This will require the user
to notify the customer’s IT department when the
anomaly occurs so that trace can be stopped ensuring
pertinent trace information is not overwritten. A
pager or other notification should be sent to the
Avaya Services Engineer to alert them that the capture
has been obtained.
If it has been established that the audio quality
problem occurs on trunk calls, it is recommended that
a trunk ID button be programmed on the user’s phone so
that when the audio anomaly occurs, the specific trunk
accessed can be obtained.
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If the audio quality problem is reproducible,
establish a call with the phone experiencing the audio
issue and note the call traversal (the path the call
takes through the system). Once the call flow has been
identified, all the network components involved are
now known. At this point, try to determine whether the
problem is limited to specific system components by
initiating new calls that traverse a fewer number of
system components. For instance, test if the audio
issue can be reproduced on a station to station call
between two DCP Telephones on the same port network.
If the audio anomaly occurs when IP Telephones are
used, determine whether the anomaly exists with IP to
IP direct (shuffling) or whether the anomaly only
exists when calls traverse a Media Gateway VoIP
resource3 or only over a trunk call.
6. AUDIO QUALITY ANOMALIES
Having established the path the call traverses when
the customer audio anomaly occurs, combine this
knowledge with the definitions in Appendix A to
determine what system component is causing the problem
or if further measurements are required. The following
sections describe the three major categories of audio
quality anomalies (distortion, noise, and echo) and
the typical causes.
Distorted Speech
Speech distortion(s) are typically caused by low-rate
coders, improperly matched hybrid trunk circuits, and
stutter or speech-level pumping. Audio distortion
(amplitude clipping) coming from outside the
customer’s network is typically cause by the far-end
phone.
3
Note that taking a trace to an IP Phone while an anomaly is occurring is a quick
test to identify if the problem is in the phone or not by listening to the RTP
stream and comparing that to what is heard in the phone.
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Distortion from inside the customer network could be
caused by, for example, a headset user with the
microphone too close to their mouth and overdriving
the microphone circuitry.
1. Distorted Speech can be caused by a damaged
microphone in the handset, headset, or speakerphone
circuit. Swap out the equipment to determine if this
is the cause of the distortion. Use of Low-rate
coders can cause muffled speech, synthetic,
mechanical or robotic speech when network
impairments are present. Take network QoS
measurements and note the values of the typical
characteristics (packet loss, delay and jitter).
•
If the user experiences any of theses types of
speech distortions while on a call using the
4610/20/21/22 IP Phones, the user (or Services
engineer) may press the Network Audio Quality
menu item (hit the Options button and select
Network Audio Quality) to get an instant
measurement of packet loss, delay and jitter
being experienced by that phone.
•
If the problem is intermittent, working with
the customer to set up a network analyzer to
monitor the network 24/7 such that when the
user experiences the problem, the trace can be
stopped and RTP analyzed. Use this data in
conjunction with the network topology and call
flow to determine what network element is
causing the anomaly.
2. Amplitude clipping can be determined by running an
Ethereal trace and converting the RTP stream to a
.wav file and analyzing that file with a tool such
as Audacity. The clipped speech signal can be
observed graphically with that tool.
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If amplitude clipping is present, identify solution
elements that add gain and verify each element is
applying the expected amount of gain.
3. Clipping can be caused by packet loss which can be
determined using a network analyzer. Also, e-can
settings 1 thru 4 on the TN464GP boards are very
aggressive and can cause clipping. Administering the
DS1 (GP) e-can setting to 8 or 9 can
reduce/eliminate the clipping4. On DS1 HP boards, ecan setting 3 is a less aggressive echo canceller
and can reduce/eliminate clipping.
4. Clipping during double-talk can be caused by the
suppressor function of an echo canceller in the
network. DS1, MedPro/Cruiser, Crossfire, and ATM all
use e-cans with suppressor functions.
5. Stutter can be traced to it’s origin by setting up a
network analyzer to analyze the call topology and
where the bottleneck is occurring.
6. Speech-level pumping can be caused by the DS1 e-can
setting 4 (on GP cards) or by the telephone’s AGC.
Adjusting the GP type DS1 e-can setting to 7 or 8 or
turning off AGC are fixes for this problem. Also,
high transmit gain levels can cause pumping.
Reducing the TX gain in the 46xx Settings file use
by the IP Telephones can also reduce/eliminate
pumping5.
4
Changing the e-can setting affects active calls. For this
reason, it’s beneficial to change he e-can setting while the
audio anomaly is occurring in order to get immediate feedback on
the change.
5
This should only be performed by Tier IV or under Tier IV
supervision.
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Noise
Noise occurs in most telephones at low levels. It’s
only when the levels are perceptible that complaints
arise. The following provides noise types and ways in
which to capture information in order to analyze and
pinpoint the cause of the noise.
1. Hiss or white noise is common at low levels.
Requirements allow for up to 20dBrnC of idle-channel
noise to appear in the receiver of a handset or
headset. Typical idle-channel noise should measure
approximately 15dBrnC. This can be found using a
pair of Sage 925VSTs. Or a Sage 925VST with a Sage
Responder.
2. Static causes can be identified by using a network
analyzer to determine if there is packet loss or bit
errors.
3. Motor-boating or Hum can be caused by power-line
interference. Checking grounds and any nearby high
voltage sources can uncover the cause for this
problem.
4. Distorted Music-on-Hold is typically caused by a
low-rate coder. If so, changing the coder will fix
this problem.
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Echo
Echo, either acoustic or electric, is a function of
delay in the call path6. The steps below address how to
determine the cause of the delay.
1. Speech back to the far-end. If the far-end no longer
complains of echo after muting the near-end phone,
then this confirms the near-end is the source of the
echo. If the near-end is speakerphone, verify the
environment setup. If near-end headset user, verify
proper cords used. If headset with M12 amplifier,
verify the amplifier settings and proper cords are
used. Taking a trace on an IP Phone while echo is
heard can identify if the echo is in the phone or
from the far-end (if no echo is heard in the trace,
then it’s in the phone).
2. Electrical echo (constant, intermittent, residual,
distorted) is either at the analog trunk or in the
PSTN. Terminating the trunk locally and testing for
echo will identify if the source of echo is in local
or in the PSTN. Verify DS1 e-can is being utilized
and not the MedPro e-can resources.
7. OTHER INFORMATION
It is highly recommended that you do not adjust the
Loss Plan or Terminal Parameter gain settings. This is
not a recommended option for addressing audio
impairments.
For optimum headset performance, use only Avaya
headsets best suited for the environment7.
6
For more in-depth echo discussion, see Echo and Echo Control
document - COMPAS ID 106541.
7
http://www.avayaheadsets.com
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8. REFERENCES
[1]
Terminology for voice-related artifacts in
telephony, CID # 114121, September, 2005.
[2]
Echo and Echo Control, CID # 106541, Issue
0.4, June, 2005.
[3]
Avaya 4600™ Series IP Telephones - Modular
Configuration Power Options, CID # 93139, Issue 1.3,
January, 2004
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APPENDIX A: Audio Processing
Components & Audio Quality
Terminology
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In a typical telephony call, speech from talker to
listener often passes through the following processing
components and in the order identified in Figure 1.
speaker
reverse of below
Echo
controller
+
-
+
Echo
controller
Expander
(noise
reduction)
Speech
encode
network
Speech
decode
Packet-loss
conealment
Automatic
Gain
control
mic
Figure 1.
The talker’s voice enters at the microphone on the
left side of Figure 1, then to the microphone
expander, voice coder, network transport, voice
decoder, packet-loss concealment, echo controller,
automatic gain control and, finally, the listener’s
ear. Brief definitions of these components are given
below.
Echo controller. This is a broad term meaning an
echo canceller, echo suppressor, or a combination of
the two. Speakerphone algorithms are also included. An
echo controller prevents a talker from hearing distant
reflections (echoes) of his/her own voice, reflections
caused by acoustic or electrical reflection points
within the telephone network and end-user equipment.
Echo controllers are often only partially successful,
and this is why echo is sometimes heard even though
the call path is known to include echo controllers.
Often, people use the “echo canceller” when in fact
what is being referred to is an echo controller.
Echo canceller. Software or hardware implementations
of a digital signal processing algorithm designed to
model and subtract-out – or cancel – the reflection,
or echo, of a speech signal. Strictly speaking, an
echo
path
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echo canceller does not introduce attenuation or
suppression into the speech paths to reduce the
loudness of echo. The term canceller refers to an
adaptive digital filter that models the physical echo
path and subtracts that (excited) model from the
return speech path.
Echo suppressor. Like echo canceller, above, except
the echo level is reduced or eliminated by applying
suppression or attenuation to the return speech
channel. The use of attenuation causes other audio
artifacts, including chopping or clipping of speech
utterances and/or pumping of the loudness level of a
caller’s speech.
Microphone expander (and or noise reduction). A
microphone expander is a traditional and relatively
simple method of improving the speech-signal-tobackground-noise ratio emanating from the microphone
path. An expander attenuates weak room background
noises while passing unaltered the relatively loud
speech of the talker addressing the handset (or
headset, or speakerphone).
Speech coder (encoder and decoder). The raw
speech signal, once digitized, is often digitally
encoded for transmission into the telephone network.
Encoding has one purpose, namely, to reduce the bitsper-second rate of transmission required to
communicate voice from one end to the other. Highly
compressive codecs, such as a G.729 codec, reduce
speech to a low transmission rate (8000 bits-persecond), but sacrifice voice quality in doing so.
Higher voice quality is experienced in systems using
the traditional G.711 codec (mu-law codec), since
G.711’s higher transmission rate of 64,000 bits-persecond better captures the nuances of speech.
Regardless of coding scheme, at the receiving side,
the speech decoder reconstructs (an approximation to)
the original speech for playback.
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Packet-loss concealment. Often combined with
speech decoders. When the network path includes
packet-speech transmission links, like VoIP, speech
packets can be lost because of network failures. In
such cases, a concealment algorithm attempts to fillin missing speech samples. Concealment can work well
when the rate of lost speech is very low, say, less
than 2% of transmissions.
Automatic gain control. Automatic gain control
devices apply signal gain or loss automatically in an
attempt to keep the speech sound level at the
listener’s ear relatively constant. Therefore, AGC
boosts low-level speech while reducing speech levels
that are too loud. Such devices have been used for
decades in audio broadcasting and recording
applications.
The following sections outline the troubleshooting
steps in order to identify audio issues.
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Speech Distortions
Distorted Speech
Speech accompanied by an unnatural buzzing or raspy
sound. A classic example of distortion occurs in the
case of a far party who is speaking too loudly or too
close to the handset or headset microphone. The far
party’s speech saturates either the mechanical or
electrical capabilities of the handset, causing
overload distortion or amplitude clipping. Other
sources of speech distortion include: defective or
improperly (impedance) matched hybrid trunk circuits,
damaged microphones, loudspeakers or earpieces (a
rarity), and low-bit-rate speech coders (VoIP and
wireless).
Muffled Speech
Speech that has an unnatural loss of high-frequency
content. Muffled speech may be caused by, for example,
poorly designed microphone assemblies in handsets (in
particular, wireless handsets) and low-bit-rate speech
coders (VoIP or wireless).
Reverberant speech (also hollowness or
speaking-in-a-tunnel effect)
Sounds like the person you are listening to is
speaking in a barrel or large empty room. This can be
the case when the talker is using a speakerphone, but
it can also be the case when there is network echo,
e.g., in a teleconference without echo control.
Also see ECHO section anomalies: hot sidetone, shortpath acoustic echo, and electrical echo.
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Synthetic, Mechanical, or Robotic Speech
This can be very subtle or very severe, or very
consistent or intermittent. In the most severe case,
the pitch information has been lost making the speech
sound monotonic and robotic. Recognizing who is
speaking is often difficult. Faulty or bad acting
hardware (e.g., a DSP chip in a cell phone, base
station, or VoIP gateway) can cause this. This may be
port/resource dependent but consistent on the faulty
port or resource. If this is occurring intermittently,
sudden bursts of packet loss (or frame erasure in a
wireless network) may be the cause as the packet-loss
concealment scheme struggles to interpolate the speech
signal. A less extreme case in which you may recognize
the speaker, but yet the speech sounds somewhat
artificial or synthetic has been known to occur as
well. Sometimes the effect includes a swirling sound
accompanying or in place of certain frequencies of the
voice signal. It may be very subtle and is often
linked to the usage of a low-bit-rate speech codec,
such as G.729 in the presence of background noise, or
possibly the occurrence of packet loss or frame
erasures, as occurs in VoIP and wireless networks.
Amplitude Clipping
See definition for distortion above.
Clipping
Portions of the speech signal are not heard. This can
occur in packet-switched networks when, for example,
large numbers of successive speech packets are not
received because of excessive network congestion.
Common in wireless phones, where the RF-signal
strength fades as the user moves within the
environment.
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Clipping during Double-talk
Clipping, as defined above, but heard only when both
parties of a telephone call talk at the same time.
When it occurs, this effect is almost always caused by
the excessive use of echo suppression (see definition)
at some point within the network. In this case,
clipping of speech utterances is not caused by lost
speech packets or, in the case of wireless phones, RF
fades, though those artifacts may also be present in
the same call.
Stutter
This is often used to describe an effect caused by
repetition of short bursts of noise or speech, such as
“da-da-da-da…” or “fa-fa-fa-fa…” Stutter distortion
can occur in packet-speech networks when one or more
network elements (e.g., router or switch) become a
bottleneck to the timely transmission of speech
packets.
Speech-level pumping
Pumping is often used to describe a varying speechloudness level, that is, were the speech gets louder,
softer, then louder again, etc., over the course of a
call, often over a period of just several seconds.
Automatic gain control devices can cause audible and
distracting pumping.
NOISE
Hiss or White Noise
Relatively natural-sounding noise containing energy at
all frequencies. Low-level, idle-channel hiss noise
can be perceived on nearly every telephone call when
no person is speaking.
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Static
Impulsive, ticking noise, similar to the sound of an
AM radio when tuned to a very weak or nonexistent
radio station. In a packet-speech network, can be
caused by lost speech packets and/or bit errors. May
also be used to describe power-line hum (see
definition below).
Motor-boating
Repetitive noise that is separate and distinct from
the talker’s voice. Motor-boat noise differs from
static in that it is repetitive or non-random. Motorboat noise can be caused by periodic disruptions
(errors) to digital speech circuits and power-line
interference at telephone endpoints.
Hum
Sounds like humming, as in “Hmmmmm…” Hum noise often
occurs when a source of 50 Hz or 60 Hz electrical
power is located near a telephone. The power source
emits an RF (radio frequency) field that induces a
hum-like noise that is heard through the phone’s
handset/headset earpiece or speakerphone loudspeaker.
Distorted Music-on-Hold
Low-bit-rate codecs such as G.729, and G.723, were
created to efficiently encode and transport speech but
not music. Thus the usage of these and other codecs
may distort and ruin the music signal. This can be
subtle or severe depending on the music source. As a
remedy you can try to experiment with various music
sources, but it would be best to use a more
appropriate codec such as G.711, G.726, or G.722.
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ECHO
There are only two physical sources of echo in
telephony: electrical echo, or network echo, and
acoustic echo. Electrical echo is caused by a
reflection of the speech signal at 2-to-4-wire hybrid
circuitry. This circuitry is present in our own analog
trunk cards, and it also exists deep in within the
PSTN (at customer premises, for example). Acoustic
echo is caused by the physical coupling (air path,
appliance-body path) between a loudspeaker and a
microphone, for example, in a speakerphone, a handset
and a headset. Whether or not a talker actually
perceives electrical or acoustic echo depends on the
loudness of his/her reflected voice signal and the
roundtrip delay that that reflection suffers. The
loudness of the reflection at the point of reflection
depends upon the electrical impedance mismatch, for
electrical echoes, and the acoustic gain of the
loudspeaker-to-microphone path, for acoustic echoes.
The roundtrip delay is a function of the path the
reflected signal traverses, which in turn is a
function of the call topology
Electrical echo, also called network echo
Reflection of a talker's speech signal at a point of
2-to-4-wire conversion caused by an impedance mismatch
at the point of analog-to-digital conversion.
Acoustic echo
Reflection of a talker's speech signal at an acoustic
endpoint caused by the acoustic coupling between the
loudspeaker and microphone.
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Constant echo
When talking, the perception of echo with every
utterance. Such cases occur when there is a physical
electrical or acoustic echo path but no echo
controller in the call topology to control echo.
Additionally, constant echo may result even though an
echo controller is known to be in the call path; this
indicates a complete failure of the echo controller,
usually because the capabilities of the echo
controller are exceeded (e.g., the echo tail length
exceeds the specifications of the echo controller).
Intermittent echo
When talking, the occasional perception of echo.
Intermittent echo often caused by the intermittent
failure of an echo controller in the call path. The
echo suppressor within the echo controller may fail to
engage (to apply echo attenuation) when necessary,
with the result that short bursts of echo become
audible. In acoustic echo control applications
(speakerphone) in which people or objects close to the
speakerphone are moving, the change to the physical
echo path often results in audible intermittent
acoustic echo to listeners at the other end of the
call.
Residual echo
When talking, the perception of very low-level (quiet)
echo. The echo could be either constant or
intermittent. Residual echo can be caused by PSTN
electrical echo that is not entirely removed by the
echo controller in the call path.
Distorted or buzz-like echo
When talking the perception of a distorted echo or
buzz-like sound. This can be caused by a non-linear
echo source. An example of this is saturation
distortion at an analog trunk interface. In this case,
signals low in amplitude are reflected cleanly, but
signals high in amplitude are returned with
significant distortion making it difficult for an echo
canceller to control echo.
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Such distorted echo can be perceived constantly or
intermittently, depending on the degree of distortion
and the echo canceller(s) involved.
Slapback or kickback acoustic echo
This is strictly a phenomenon of acoustic echo. With
speakerphones, slapback or kickback echo is the
intermittent echo perceived at the ends of one's
utterances. This can occur with both older-model halfduplex speakerphones and newer-model acoustic-echo
canceling speakerphones. For example, a talker
speaking into a handset utters the phrase “Please send
me the check” and perceives echo primarily at the end
of his/her sentence. This echo is described as hearing
just the sound “eck” or “k” of the word “check,” or as
a slapping sound such as that made by slapping one’s
palm against a desktop. Commonly, slapback/kickback
echo is caused by acoustically reverberant rooms.
Large offices and conference rooms can have long
reverberation times. In such rooms, the speakerphone
senses at its microphone a reverberated version of the
word “check” (our prior example) several tens or even
hundreds of milliseconds after the far talker has
finished saying the word “check.” The speakerphone
algorithm detects this reverberated speech at its
microphone, detects no speech at its receive-path
driving the loudspeaker, and decides to transition to
transmit mode. The reverberated version of "check" is
transmitted back to the far talker, where it is
perceived as echo.
Whether a slapback reflection occurs depends on the
speakerphone acoustics, speakerphone algorithm and
room acoustics. Whether slapback echo is perceived as
echo by the far talker depends on the loudness of the
echo and the roundtrip delay. Some have described
slapback echo as speakerphone false switching, where
it is meant that the speakerphone has switched too
quickly from the receive mode to the transmit mode.
This is not actually a false switch, however; the
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speakerphone merely rapidly switches between states
because it cannot distinguish between microphone
energy that is due to a talker who is local to the
speakerphone room and energy due to reverberation of
the received speech. A speakerphone that enforces a
longer hold-time governing the smallest amount of time
for a switch between the receive and transmit states
will in general suffer less from slapback echo. A
common drawback, however, of a longer receive-totransmit hold-time is a higher perception of clipped
utterances during doubletalk; in other words, a
greater perception at both ends of the speakerphone
call that the speech path is half duplex.
Sidetone
In handsets and headsets, a portion of the microphone
energy is fed back to the earpiece so that the user of
the handset/headset does has a psychoacoustic
experience that simulates the case in which the user's
ear is not occluded by an object (the handset
earpiece). Without sidetone injection, the user
experiences the psychoacoustically bothersome
condition that can be demonstrated to oneself by
pressing a finger into one ear while speaking. With
one ear occluded, the sound of one’s own voice is
dominated by the path through the interior of the head
(skull, etc.) instead of around the head, an effect
that most people find objectionable.
Hot sidetone
In a handset or headset, microphone-to-earpiece
sidetone injection is not normally noticed. Some
digital phones, in particular, IP phones in which the
internal audio processing frame rate is 5 ms or
greater, inject sidetone with an appreciable delay
(e.g., 5 ms) in the microphone-to-earpiece signal
path. This delay causes the sidetone to sound
reverberant and/or louder than normal, or hot. Though
hot sidetone is a type of echo source – because some
people may use the term “echo” to describe hot
sidetone – it is generated local to the telephone, not
at some point within the telephone network.
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Short-path acoustic echo, short-path electrical
echo
Acoustic or electrical echo that occurs in a very
short roundtrip call topology. This type of echo is
commonly described as a hollow sound or sound of
speaking in a barrel (see 2.1.3). In a digital-todigital phone call (think DCP-to-DCP), station-tostation, the roundtrip delay is usually very small,
less than 10 ms. Some digital speakerphones produce
significant acoustic echo, which is not canceled,
suppressed, or otherwise controlled in this simple
call topology. In these cases, and depending on the
volume setting of the far-party’s speakerphone and
near-party’s listening handset, the near party may
perceive echo and refer to this as hot sidetone.
Again, this is truly acoustic echo from the
speakerphone but is returned to the talker with such a
short roundtrip delay that it is perceived as
hollowness or reverberance rather than as classic
echo. Because of the short roundtrip delay in this
case, it can be difficult to distinguish between hot
sidetone (see definition) and short-path echo.
Short-path electrical echo refers to an electrical
echo source in which the roundtrip echo path is also
short, say less than 10 ms. Again, in such cases, it
may be very difficult to distinguish between shortpath echo and hot sidetone (see definition).
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APPENDIX B: Basic IP Telephony
Solution Diagram
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Figure 2.
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